1. Field of the Invention
The invention relates to apparatus and methods for recovering alignment from a non-ideally encoded multi-bit digital signal without the need to embed framing information within the encoded signal or append framing bits thereto and particularly to recovering septet alignment in a 7 kHz audio digital signal without the need to embed any framing information within the audio digital signal itself.
2. Description of the Prior Art
Presently, voice communication over a single analog telephone loop is generally limited to a bandwidth of approximately 3.1 kHz. Inasmuch as a significant amount of the energy that occurs in human speech occurs at frequencies in excess of 3.1 kHz, 3.1 kHz bandlimited speech often sounds distorted and quite unnatural. For some time, the art has widely recognized that speech which has been bandlimited to 7 kHz is highly intelligible and appears rather natural with a noticeable improvement in quality over 3.1 kHz bandlimited speech.
As such, in certain audio communication applications where speech clarity and reduced distortion are necessary, such as illustratively in teleconferencing and providing commentary quality audio channels to interconnect remote program sources to broadcast stations, telephone channels that provide a 7.5 kHz audio bandwidth are often utilized.
Initially, one would think that 7.5 kHz channels could be easily provided through a dedicated point to point wideband analog link supplied by a local telephone company. Unfortunately, dedicated wideband analog links suffer several serious drawbacks which tend to greatly limit their utility and availability. First, these links are susceptible to noise, particularly hum resulting from power line induction. Second, since the need for such links tends to be geographically widespread and transitory in nature, each link is generally separately installed prior to the initiation of its specific use and then torn down after this use is concluded. Third, these links, primarily being copper wire in nature, are not perfect transmission media. In fact, the transmission characteristics of such a link vary with changes in the length of the link and are dependent upon the frequencies being propagated therethrough. In essence, these characteristics attenuate the peak amplitude of and impart phase distortion to the analog signal applied to the link. To assure that a wideband analog link provides a virtually flat response thereby assuring, to the extent possible, that the signal received at a far end of such a link closely replicates a source signal that is applied to a near end of the link, ancillary electronic analog circuitry needs to be inserted, typically at a far end of the link, to compensate for the actual attenuation and impedance of the link being used, i.e. so as to "align" the line. Unfortunately, this ancillary circuitry tends to be rather complex and requires a highly skilled craftsperson to first properly adjust the circuitry such that its characteristics are typically the inverse of those of a given link and subsequently operate the circuitry to maintain the link in an aligned condition in response to any changes in the frequency content of the signal carried thereover. Moreover, a telephone company, even one in a large metropolitan area, often has only a very small number of such craftspersons. Hence, due to the susceptibility of wideband analog links to noise, the need to expend resources to separately install and tear down each link, and the need to utilize highly skilled personnel to align and maintain each link during its use, wideband analog links tend to be widely disfavored by the local telephone companies for use in many applications and for that reason are seldom used.
Given the drawbacks associated with wideband analog links, the art has instead increasingly turned to converting a high quality audio signal into digital form and carrying the digitized signal over a digital channel to provide a so-called digital program channel. Several reasons account for increasing use of a digital signal. First, digital links are immune to noise, such as power line induction and are widely available as part of existing T carrier systems. In addition, digital data is unaffected by any change in the impedance of a physical link that carries a digital signal. As such, given the advantages afforded by digital transmission, digital program channels are widely used today to carry high quality audio signals.
A digital program channel has been and, to a certain extent, is still widely implemented in the following basic manner. A high quality analog audio signal is routed, via a four wire link, to a digital channel bank, typically located in a central office, where the analog audio signal is first bandlimited to 7.5 kHz. The resulting bandlimited signal is then sampled at 16 kHz with each sample then being digitized into 14 bit form through an analog-to-digital (A/D) converter. Each resulting 14 bit digital word is then converted through instantaneous A-law companding into an 11 bit pulse code modulated (PCM) word. A single parity bit is then added to each resulting word to form a twelve bit data word for each sample. Given a 16 kHz sampling rate, the resulting transmission speed becomes 192 kbit/second. Unfortunately, transmitting a signal at this bit rate requires the use of three adjacent 64 kbit/second digital channels that appear on a digital (D) channel bank. Moreover, since an analog loop, preferably unloaded, is still required between a customer premise and the central office, noise, such as hum, can be and is often disadvantageously induced into the analog signal prior to its conversion into digital form thereby injecting noise into the digital signal. In addition, since an analog loop along with various attendant services, such as equalization, that assure its proper operation, is required, the installation of a 192 kbit/second 7.5 kHz digital program channel is still rather costly.
In an effort to decrease the number of 64 kHz digital channels from three to one that is needed to transmit a high quality digitized audio signal and also eliminate the need for any analog loops by locating the digital encoder at a customer premise rather than at a central office, the art has turned to transmitting digital audio signals using adaptive differential pulse code modulation (ADPCM) techniques. Inasmuch as the ADPCM circuitry, which is owned by the customer, is located in the customer premise and directly (i.e. DC) coupled to the incoming audio signal, analog loops are eliminated with the resultant ADPCM digital signal being substantially noise free with an essentially flat frequency response. Using such a signal advantageously simplifies the responsibility of a local telephone company to merely provide an end-to-end 64 kbit/second digital channel.
With ADPCM techniques, a high quality audio signal is applied through a suitable low pass filter to bandlimit the signal to 7 kHz. The resulting bandlimited signal is then sampled at 16 kHz, with each sample being converted into 15 bit form. Using adaptive differential techniques, an encoder reduces each sample to 8 bits at a reduced sample rate of 8 kHz. In particular, given a present sample, the encoder predicts the value of a successive sample, through use of a set of parameters obtained from processing previous samples, then subtracts the predicted sample value from the actual sample value to produce a difference value and finally transmits the difference value. Adaptation occurs inasmuch as a scale that is used to encode the difference value adaptively varies according the magnitude of prior difference values. Inasmuch as most of the available energy contained in a speech signal lies within a 0-4 kHz frequency sub-band with the remainder residing in a 4 kHz -8 kHz sub-band, an ADPCM encoded 4-bit signal disadvantageously possesses differing ratiometric values of signal to distortion for each of the two sub-bands. As such, the art has turned to encoding a 7 kHz audio signal using sub-band ADPCM, as described in detail in "7kHz Audio Coding Within 64 kbit/s", CCITT Recommendation G.722, IXth Plenary Assembly Melbourne 1988, Study Group XVIII--Report R 27, Document AP IX-142-E, July 1988, pages 231--318; (henceforth referred to as the CCITT G.722 Recommendation). In essence, sub-band ADPCM first relies on sampling an incoming 7 kHz audio signal at 16 kHz with a 15 bit resolution. Then, through the use of two linear-phase non-recursive digital filters (i.e. a quadrature mirror filter), the sampled digital signal is split into two bands: a lower 0-4kHz sub-band and an upper 4 kHz-8 kHz sub-band. Thereafter, the lower sub-band is encoded into a 6-bit 8 kHz ADPCM signal while the upper sub-band is encoded into a 2-bit 8 kHz ADPCM signal. The resulting eight bits are then multiplexed together to form an 8-bit 8 kHz ADPCM signal for transmission over a single 64-kHz digital channel. Each 8-bit word is commonly referred to as an octet. Sub-band ADPCM advantageously provides a coded signal that, through the use of differing amounts of bits for each sub-band, evenly spreads signal distortion across the entire 0-8 kHz band.
As such, sub-band ADPCM now appears to be a favored coding technique for transmitting 7 kHz audio signals on a single 64 kbit/second digital channel.
Unfortunately, in North America, most digital channels that are currently available for user access have been designed to carry data and not voice at a maximum user capacity of 56 kbit/second. In particular, while these digital channels do in fact carry 64 kbits/second, only seven bits in each octet are available to carry user supplied data at a rate of 56 kbit/second with the eighth bit, which is always fixed to a one, being reserved for telephone company use in extracting timing information from the bit stream appearing on the channel. For that reason, these channels will be henceforth referred to as 56 kbit/second channels. As such, these channels do not possess the requisite capacity to carry the octets that form the 7 kHz ADPCM encoded audio digital signal. However, the sub-band ADPCM coding technique, as set forth in the CCITT G.722 Recommendation, advantageously possesses the flexibility to permit a user not to transmit the least significant bit of every octet with only minimal degradation occurring in the quality of the audio signal that has been reconstructed at a far end location. As such, every sub-band ADPCM encoded octet can become a 7-bit word (i.e. a septet), which, in turn, reduces the bit rate of the coded words themselves to essentially match that of an available 56 kbit/second channel. Unfortunately, a 56 kbit/second channel is not able to carry a clocking (framing) bit in addition to each septet. This deficiency is irrelevant for data inasmuch as a 56 kbit/second channel was designed to merely provide a bit based transport media which is all that is required to carry data. As such, the user of such a channel rather than its provider, i.e. the local telephone companies, has the responsibility to provide necessary framing by embedding suitable protocols within a data stream carried on the channel using terminal equipment connected to both ends of the channel. However, whenever these channels are to be used to carry audio information, the equipment connected to these channels requires that some form of framing be provided in order for that equipment to properly group the bits of each septet.
One technique proposed in the art for providing septet alignment involves multiplexing framing information onto each septet. See, specifically, that disclosed in "Frame Structure for a 64 kbit/s Channel in Audiovisual Applications Teleservices," CCITT Recommendation H.221, IXth Plenary Assembly Melbourne 1988, Study Group XV, Report R 33, Document AP IX-59-E, April 1988, pages 104-121 (hereinafter referred to as the CCITT H.221 Recommendation). Here, this technique injects framing information into one bit of each septet, typically bit seven. As a result of this approach, this technique has several drawbacks. First, this technique reduces the available audio bit rate to 48 kbit/second. Second, both the receiving and transmitting terminal equipment that are situated on opposite ends of a 56 kbit/second digital channel must use the same framing technique. Third, unrecoverable framing errors could occur if the terminal equipment situated at both ends of the channel are configured to exploit the full 56 kbit/second bit rate of the channel to transmit audio information. Specifically, with this technique the terminal equipment situated at both ends of the channel may be configured to drop framing, i.e. cease transmitting framing information, after septet alignment has been obtained. Unfortunately, if a failure (e.g. a slip on the channel) occurs and framing is lost, then the receiving terminal equipment is required to detect that loss, and subsequently transmit a control signal back to the transmitting terminal equipment to re-instate the framing signal and re-start an alignment procedure. This can occur only if the channel provides two way communication and special purpose communication circuitry exists within the terminal equipment situated at both ends of the channel to provide special signalling procedures therebetween. In the absence of satisfying either of these requirements, framing errors would likely be unrecoverable. In view of these drawbacks, the technique presented in the CCITT H.221 Recommendation simply does not provide a satisfactory solution for providing septet alignment for a 7 kHz digital audio signal transmitted over a 56 kbit/second digital channel.
Therefore, a need exists in the art for a method and accompanying apparatus for providing septet alignment for a 7 kHz sub-band encoded ADPCM digital audio signal without embedding framing bits within this signal. This alignment method should be sufficiently robust to enable framing to be quickly re-established after a loss of alignment occurs. Moreover, this method should not require the use of special signalling procedures, thereby eliminating the need to incorporate special purpose communication circuitry in both the receiving and transmitting terminal equipment that are connected to opposite ends of a 56 kbit/second digital channel.